VoIP and its associated control protocols such as Session Initiation Protocol (SIP) and H.323 is a viable mechanism for transmitting real-time voice over digital data circuits. With SIP and a proxy server, load can be shared among parallel network elements. Two common problems for VoIP calls are: a failure when there is no proxy server to handle the new inbound call; and the failure of a network, or a network element, during a call. When the latter case happens, the voice connection is broken and typically the caller hangs up after hearing nothing for a period of time.
Another point of weakness in a VoIP solution is the gateway. It is the interface to the PSTN connection, and if it fails, then all calls through the gateway will be lost. Typically the larger the gateway the better the economics of the cost per voice circuit, so the customer typically buys “larger” gateways. This expands the scale of the problem when a gateway fails.
Contact Centers typically require a very high availability of the voice media channel. In time division multiplex TDM based voice systems in common use in the call center today, various redundancy schemes prevent the failure of single parts of the hardware from affecting new calls, although they will typically cause a failure of the calls that went through the network element that failed, causing the calling contact to be disconnected. When a contact has been waiting in queue and experiences such technical difficulties it will typically lead to serious customer dissatisfaction and probable customer service issues for the Contact Center operator, such as lost sales, lost customers, and abused agents.